This invention relates generally to voice over Internet protocol (VoIP) technology, and particularly to how call routing is performed to improve bandwidth efficiency, call quality, call setup time and call handling flexibility.
In a public switched telephone network (PSTN), an initiating phone connects to a circuit switch and the PSTN via a first POTS (Plain Old Telephone System) line. A destination phone connects to the circuit switch and the PSTN via a second POTS line. The circuit switch electrically connects the initiating phone to the destination phone over the PSTN. The electrical connection is maintained for the entire duration of a phone call between the initiating phone and the destination phone. The electrical connection in the PSTN is commonly referred to as “circuit switched.”
Voice over Internet protocol (VoIP) is a technology that permits phone calls to be carried over the Internet as opposed to over the PSTN. Being packet switched, carrying voice over the Internet is far more bandwidth efficient than circuit switched voice communications. In VoIP, a device known as an analog telephone adapter (ATA) or media gateway serves as an interface between an analog phone and the packet-based Internet. The ATA may be a standalone device or may be incorporated into another device such as a cordless phone base station or broadband modem. An initiating ATA converts analog signals from an initiating phone into packets using a voice codec (coder/decoder) algorithm. At the destination phone, to receive an incoming VoIP phone call, a destination ATA receives packets into a buffer and uses the same codec algorithm to convert the packets back into analog signals.
Typically, ATAs provide VoIP functionality via a connection to a broadband modem, such as through a cable modem or a digital subscriber line (DSL) connection to the Internet. Typically the VoIP service provider strives to emulate the behavior and reliability of the PSTN while offering a lower cost for delivering the service or increased functionality.
In a typical call scenario a call is initiated by a VoIP subscriber by going off hook from a telephone connected to an ATA. The ATA collects the DTMF tones generated by the analog phone and determines which phone number the subscriber is calling. A message with the target phone number is sent over the broadband network connected to the ATA to a server which determines how the call should be routed to complete the call. When the call is directed to a POTS subscriber, the packets associated with the call are ultimately routed to a termination gateway which converts the packets from packetized digital format to an analog signal for eventual delivery to the destination phone number. If the call is directed to another subscriber to the same VoIP service then the call may be routed directly to that subscriber's ATA, bypassing the termination gateway. If, as described in U.S. patent application Ser. No. 10/888,603 for “Connecting a VoIP phone call using a shared POTS line” (U.S. Pub. No. 2006/0007915), a PSTN line is connected to the ATA, then the call may be completed through that PSTN connection.
Previous VoIP and packet-based telephony systems have been hampered by several difficulties related to call routing which degraded the quality, reliability, flexibility and bandwidth efficiency of calls they carried. In general, these problems have been accepted as an inevitable trade-off for the advantages (e.g., low cost, portability,) VoIP systems provide. Since the underlying technology of VoIP systems is likely to improve, it is assumed that the present problems with VoIP systems will diminish over time.